[AsteriskBrasil] SIP Trunk Asterisk x Alcatel

Sexta Fevereiro 22 09:49:12 BRT 2013

Bom dia,

   Estamos participando da integração de um asterisk 1.6.2.11 com uma 
Alcatel via SIP. Quando ligo do asterisk para a Alcatel recebo um 
forwarding da chamada de volta para o asterisk. Segue maiores 
informações:

sip.conf:

[alcatel]
host=172.16.1.3
context=from-Alcatel
type=friend
nat=no
disallow=all
allow=alaw

extensions.conf

exten => _6X.,1,Dial(SIP/${EXTEN:1}@alcatel)
  same => n,HangUp

no console do asterisk durante a chamada

     -- Executing [69202 em saida:1] Dial("SIP/jefferson-00001a43", 
"SIP/9202 em alcatel") in new stack
   == Using SIP RTP CoS mark 5
     -- Called 9202 em alcatel

     -- Now forwarding SIP/jefferson-00001a43 to 
'Local/9202 em from-Alcatel' (thanks to SIP/alcatel-00001a44)

[Feb 22 09:36:24] NOTICE[19305]: chan_local.c:538 local_call: No such 
extension/context 9202 em from-Alcatel while calling Local channel
[Feb 22 09:36:24] NOTICE[19305]: app_dial.c:789 do_forward: Failed to 
dial on local channel for call forward to '9202 em from-Alcatel'
   == Everyone is busy/congested at this time (1:0/0/1)
     -- Executing [69202 em saida:2] Hangup("SIP/jefferson-00001a43", "") 
in new stack
   == Spawn extension (TI, 69202, 2) exited non-zero on 
'SIP/jefferson-00001a43'

Segue sip debug deste peer

asterisk*CLI> sip set debug peer alcatel
SIP Debugging Enabled for IP: 172.16.1.3:5060
   == Using SIP RTP CoS mark 5
     -- Executing [69202 em TI:1] Dial("SIP/jefferson-00001a46", 
"SIP/9202 em alcatel") in new stack
   == Using SIP RTP CoS mark 5
Audio is at 172.16.200.92 port 5404
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.16.1.3:5060:
INVITE sip:9202 em 172.16.1.3 SIP/2.0
Via: SIP/2.0/UDP 172.16.200.92:5060;branch=z9hG4bK7b79bc0e;rport
Max-Forwards: 70
 From: "jefferson" <sip:jefferson em 172.16.200.92>;tag=as1b46e101
To: <sip:9202 em 172.16.1.3>
Contact: <sip:jefferson em 172.16.200.92>
Call-ID: 7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Date: Fri, 22 Feb 2013 12:38:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 303517300 303517300 IN IP4 172.16.200.92
s=Asterisk PBX 1.6.2.11
c=IN IP4 172.16.200.92
t=0 0
m=audio 5404 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called 9202 em alcatel

<--- SIP read from UDP:172.16.1.3:5060 --->
SIP/2.0 100 Trying
To: <sip:9202 em 172.16.1.3>
 From: "jefferson" <sip:jefferson em 172.16.200.92>;tag=as1b46e101
Call-ID: 7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92
CSeq: 102 INVITE
Via: SIP/2.0/UDP 
172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:172.16.1.3:5060 --->
SIP/2.0 100 Trying
To: <sip:9202 em 172.16.1.3>
 From: "jefferson" <sip:jefferson em 172.16.200.92>;tag=as1b46e101
Call-ID: 7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92
CSeq: 102 INVITE
Via: SIP/2.0/UDP 
172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:172.16.1.3:5060 --->
INVITE sip:9202 em 172.16.200.92:5060 SIP/2.0
Record-Route: <sip:172.16.1.3;lr;transport=UDP>
Via: SIP/2.0/UDP 
172.16.1.3:5060;branch=z9hG4bK467b772a9f9ef266e7d514eede2bf348cc48719d6f59fafe0690167af90cbb1d
Via: SIP/2.0/UDP 
172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060
Max-Forwards: 69
 From: "jefferson" <sip:jefferson em 172.16.200.92>;tag=as1b46e101
To: <sip:9202 em 172.16.1.3>
Contact: <sip:jefferson em 172.16.1.3:5060>
Call-ID: 7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Date: Fri, 22 Feb 2013 12:38:28
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
Supported: replaces,timer
Content-Type: application/sdp
Content-Length: 236
Session-Expires: 1800

v=0
o=root 303517300 303517300 IN IP4 172.16.1.3
s=Asterisk PBX 1.6.2.11
c=IN IP4 172.16.1.3
t=0 0
m=audio 5404 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------->
--- (17 headers 11 lines) ---

<--- Transmitting (no NAT) to 172.16.1.3:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
172.16.1.3:5060;branch=z9hG4bK467b772a9f9ef266e7d514eede2bf348cc48719d6f59fafe0690167af90cbb1d;received=172.16.1.3
Via: SIP/2.0/UDP 
172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060
Record-Route: <sip:172.16.1.3;lr;transport=UDP>
 From: "jefferson" <sip:jefferson em 172.16.200.92>;tag=as1b46e101
To: <sip:9202 em 172.16.1.3>
Call-ID: 7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO
Supported: replaces, timer
Contact: <sip:jefferson em 172.16.200.92>
Content-Length: 0

<------------>
     -- Now forwarding SIP/jefferson-00001a46 to 
'Local/9202 em from-Alcatel' (thanks to SIP/alcatel-00001a47)
[Feb 22 09:38:28] NOTICE[19309]: chan_local.c:538 local_call: No such 
extension/context 9202 em from-Alcatel while calling Local channel
[Feb 22 09:38:28] NOTICE[19309]: app_dial.c:789 do_forward: Failed to 
dial on local channel for call forward to '9202 em from-Alcatel'
Scheduling destruction of SIP dialog 
'7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92' in 32000 ms (Method: 
INVITE)
Reliably Transmitting (no NAT) to 172.16.1.3:5060:
CANCEL sip:9202 em 172.16.1.3 SIP/2.0
Via: SIP/2.0/UDP 172.16.200.92:5060;branch=z9hG4bK7b79bc0e;rport
Max-Forwards: 70
 From: "jefferson" <sip:jefferson em 172.16.200.92>;tag=as1b46e101
To: <sip:9202 em 172.16.1.3>
Call-ID: 7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0

---
Scheduling destruction of SIP dialog 
'7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92' in 32000 ms (Method: 
INVITE)
   == Everyone is busy/congested at this time (1:0/0/1)
     -- Executing [69202 em TI:2] Hangup("SIP/jefferson-00001a46", "") in 
new stack
   == Spawn extension (TI, 69202, 2) exited non-zero on 
'SIP/jefferson-00001a46'

<--- SIP read from UDP:172.16.1.3:5060 --->
SIP/2.0 200 OK
To: <sip:9202 em 172.16.1.3>
 From: "jefferson" <sip:jefferson em 172.16.200.92>;tag=as1b46e101
Call-ID: 7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 
172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060
Content-Length: 0
<------------->

-- 
[]'s Jefferson B. Limeira
jbl em internexxus.com.br
(41) 9928-8628